The present invention relates generally to the field of digital communications, and, more particularly, to the selection of code points for digital transmission of information.
The demand for remote access to information sources and data retrieval, as evidenced by the success of services such as the World Wide Web, is a driving force for high-speed network access technologies. The public switched telephone network (PSTN) offers standard voice services over a 4 kHz bandwidth. Traditional analog modem standards generally assume that both ends of a modem communication session have an analog connection to the PSTN. Because data signals are typically converted from digital to analog when transmitted towards the PSTN and then from analog to digital when received from the PSTN, data rates may be limited to 33.6 kbps as defined in the V.34 Transmission Recommendation developed by the International Telecommunications Union (ITU).
The need for an analog modem may be eliminated, however, by using the basic rate interface (BRI) of the Integrated Services Digital Network (ISDN). A BRI offers end-to-end digital connectivity at an aggregate data rate of 160 kbps, which is comprised of two 64 kbps B channels, a 16 kbps D channel, and a separate maintenance channel. ISDN offers comfortable data rates for Internet access, telecommuting, remote education services, and some forms of video conferencing. ISDN deployment, however, has generally been very slow due to the substantial investment required of network providers for new equipment. Because the ISDN is not very pervasive in the PSTN, the network providers have typically tariffed ISDN services at relatively high rates, which may be ultimately passed on to the ISDN subscribers. In addition to the high service costs, subscribers must generally purchase or lease network termination equipment to access the ISDN.
While most subscribers do not enjoy end-to-end digital connectivity through the PSTN, the PSTN is nevertheless mostly digital. Typically, the only analog portion of the PSTN is the phone line or local loop that connects a subscriber or client modem (e.g., an individual subscriber in a home, office, or hotel) to the telephone company""s central office (CO). Local telephone companies have been replacing portions of their original analog networks with digital switching equipment. Nevertheless, the connection between the home and the CO has been the slowest to change to digital as discussed in the foregoing with respect to ISDN BRI service. A recent data transmission recommendation issued by the ITU, known as V.90, takes advantage of the digital conversions that have been made in the PSTN. By viewing the PSTN as a digital network, V.90 technology can accelerate data downstream from the Internet or other information source to a subscriber""s computer at data rates of up to 56 kbps, even when the subscriber is connected to the PSTN via an analog local loop.
To understand how the V.90 Recommendation achieves this higher data rate, it may be helpful to briefly review the operation of V.34 analog modems. V.34 modems are generally optimized for a configuration in which both ends of a communication session are connected to the PSTN by analog lines. Even though most of the PSTN is digital, V.34 modems treat the network as if it were entirely analog. Moreover, the V.34 Recommendation assumes that both ends of the communication session suffer impairment due to quantization noise introduced by analog-to-digital converters. That is, the analog signals transmitted from the V.34 modems are sampled at 8000 times per second by a codec upon reaching the PSTN with each sample being represented or quantized by an eight-bit pulse code modulation (PCM) codeword. The codec uses 256, non-uniformly spaced, PCM quantization levels defined according to either the xcexc-law or A-law companding standard (i.e., the ITU G.711 Recommendation)
Because the analog waveforms are continuous and the binary PCM codewords are discrete, the digits that are sent across the PSTN can only approximate the original analog waveform. The difference between the original analog waveform and the reconstructed quantized waveform is called quantization noise, which limits the modem data rate.
While quantization noise may limit a V.34 communication session to 33.6 kbps, it nevertheless affects only analog-to-digital conversions. The V.90 standard relies on the lack of analog-to-digital conversions in the downstream path, outside of the conversion made at the subscriber""s modem to enable transmission at 56 kbps.
The general environment for which the V.90 standard has been developed is depicted in FIG. 1. An Internet Service Provider (ISP) 22 is connected to a subscriber""s computer 24 via a V.90 digital server modem 26, through the PSTN 28 via digital trunks (e.g., T1, E1, or ISDN Primary Rate Interface (PRI) connections), through a central office switch 32, and finally through an analog loop to the client""s modem 34. The central office switch 32 is drawn outside of the PSTN 28 to better illustrate the connection of the subscriber""s computer 24 and modem 34 into the PSTN 28. It should be understood that the central office 32 is, in fact, a part of the PSTN 28. The operation of a communication session between the subscriber 24 and an ISP 22 is best described with reference to the more detailed block diagram of FIG. 2.
Transmission from the server modem 26 to the client modem 34 will be described first. The information to be transmitted is first encoded using only the 256 PCM codewords used by the digital switching and transmission equipment in the PSTN 28. These PCM codewords are transmitted towards the PSTN by the PCM transmitter 36 where they are received by a network codec. The PCM data is then transmitted through the PSTN 28 until reaching the central office 32 to which the client modem 34 is connected. Before transmitting the PCM data to the client modem 34, the data is converted from its current form as either xcexc-law or A-law companded PCM codewords to pulse amplitude modulated (PAM) voltages by the codec expander (digital-to-analog (D/A) converter) 38. These PAM voltage levels are processed by a central office hybrid 42 where the unidirectional signal received from the codec expander 38 is transmitted towards the client modem 34 as part of a bidirectional signal. A second hybrid 44 at the subscriber""s analog telephone connection converts the bidirectional signal back into a pair of unidirectional signals. Finally, the analog signal from the hybrid 44 is converted into digital PAM samples by an analog-to-digital (A/D) converter 46, which are received and decoded by the PAM receiver 48. Note that for transmission to succeed effectively at 56 kbps, there must be only a single digital-to-analog conversion and subsequent analog-to-digital conversion between the server modem 26 and the client modem 34. Recall that analog-to-digital conversions in the PSTN 28 may introduce quantization noise, which may limit the data rate as discussed hereinbefore. The A/D converter 46 at the client modem 34, however, may have a higher resolution than the A/D converters used in the analog portion of the PSTN 28(e.g., 16 bits versus 8 bits), which results in less quantization noise. Moreover, the PAM receiver 48 needs to be in synchronization with the 8 kHz network clock to properly decode the digital PAM samples.
Transmission from the client modem 34 to the server modem 26 follows the V.34 data transmission standard. That is, the client modem 34 includes a V.34 transmitter 52 and a D/A converter 54 that encode and modulate the digital data to be sent using techniques such as quadrature amplitude modulation (QAM). The hybrid 44 converts the unidirectional signal from the digital-to-analog converter 54 into a bidirectional signal that is transmitted to the central office 32. Once the signal is received at the central office 32, the central office hybrid 42 converts the bidirectional signal into a unidirectional signal that is provided to the central office codec. This unidirectional, analog signal is converted into either xcexc-law or A-law companded PCM codewords by the codec compressor (A/D converter) 56, which are then transmitted through the PSTN 28 until reaching the server modem 26. The server modem 26 includes a conventional V.34 receiver 58 for demodulating and decoding the data sent by the V.34 transmitter 52 in the client modem 34. Thus, data is transferred from the client modem 34 to the server modem 26 at data rates of up to 33.6 kbps as provided for in the V.34 standard.
The V.90 standard offers increased data rates (e.g., data rates up to 56 kbps) in the downstream direction from a server to a subscriber or client. Upstream communication still takes place at conventional data rates as provided for in the V.34 standard. Nevertheless, this asymmetry may be particularly well suited for Internet access. For example, when accessing the Internet, high bandwidth is most useful when downloading large text, video, and audio files to a subscriber""s computer. Using V.90, these data transfers can be made at up to 56 kbps. On the other hand, traffic flow from the subscriber to an ISP consists mainly of keystroke and mouse commands, which are readily handled by the conventional rates provided by V.34.
As described above, the digital portion of the PSTN 28 transmits information using eight-bit PCM codewords at a frequency of 8000 Hz. Thus, it would appear that downstream transmission should take place at 64 kbps rather than 56 kbps as defined by the V.90 standard. While 64 kbps is a theoretical maximum, several factors prevent actual transmission rates from reaching this ideal rate. First, even though the problem of quantization error has been substantially eliminated by using PCM encoding and PAM for transmission, additional noise in the network or at the subscriber premises, such as non-linear distortion and crosstalk, may limit the maximum data rate. Furthermore, the xcexc-law or A-law companding techniques do not use uniform PAM voltage levels for defining the PCM codewords. The PCM codewords representing very low levels of sound have PAM voltage levels spaced close together. Noisy transmission facilities may prevent these PAM voltage levels from being distinguished from one another thereby causing loss of data. Accordingly, to provide greater separation between the PAM voltages used for transmission, not all of the 256 PCM codewords are used.
It is generally known that, assuming a convolutional coding scheme, such as trellis coding, is not used, the number of symbols required to transmit a certain data rate is given by Equation 1:
bps=Rslog2N5xe2x80x83xe2x80x83EQ. 1
where bps is the data rate in bits per second, Rs is the symbol rate, and Ns is the number of symbols in the signaling alphabet or constellation. To transmit at 56 kbps using a symbol rate of 8000, Equation 1 can be rewritten to solve for the number of symbols required as set forth below in Equation 2:
Ns=256000/8000=128xe2x80x83xe2x80x83EQ. 2
Thus, the 128 most robust codewords of the 256 available PCM codewords are chosen for transmission as part of the V.90 standard.
The V.90 standard, therefore, provides a framework for transmitting data at rates up to 56 kbps provided the network is capable of supporting the higher rates. The most notable requirement is that there can be at most one digital-to-analog conversion and subsequent analog-to-digital conversion in the path. Nevertheless, other digital impairments, such as robbed bit signaling (RBS) and digital mapping through PADs, which results in attenuated signals, may also inhibit transmission at V.90 rates. Communication channels exhibiting non-linear frequency response characteristics are yet another impediment to transmission at the V.90 rates. Moreover, these other factors may limit conventional V.90 performance to less than the 56 kbps theoretical data rate.
Because digital impairments, such as RBS and PAD, may vary from connection to connection, V.90 provides for learning the levels of the code points for the PCM codewords when a connection is established. For example, in Phase 3 of the V.90 standard, a sequence of PCM levels are sent from the server modem to the client modem. The resulting levels that are received by the client modem are used to help determine the nature of the digital portion of the telephone connection and to select appropriate code points for signal constellations used to transfer data.
During transmission and acquisition of these levels by the client modem, some levels may be corrupted by noise, non-linearities, and other impairments within the network. Large disturbances in the levels acquired by the client modem can significantly detract from the client modem capability to select appropriate code points for its signal constellations, which may lead to sub-optimum connections and possibly failure to connect in some cases.
The digital PSTN transports information using a six symbol framing structure. That is, a frame includes six data frame intervals with each data frame interval holding a single symbol. One mechanism for reducing the impact of corruption of the levels is to average levels from different intervals and then utilize the average in constellation generation. However, because of the presence if digital impairments such as robbed-bit signaling, such averaging may result in further error. Accordingly, a need exists for improvements in reducing the impact of noise, non-linearities and other impairments on constellation generation in a PCM modem.
In light of the above discussion, it is an object of the present invention to reduce the impact that noise spikes may have on the selection of code points for a modem.
It is another object of the present invention to reduce the impact of noise spikes even in the presence of digital impairments such as robbed bit signaling.
These and other objects, advantages, and features of the present invention may be provided by averaging measured levels from differing intervals and differing code points based on their proximity with other measured levels. Incorporating measured levels for more than one code point allows for increasing the number of measured levels (samples) in the average for a code point. Thus, the averaging of the present invention may take advantage of characteristics of digital impairments such as robbed-bit signaling which may collapse the measured levels of one code point onto another to increase the number of measured levels (samples) in an average. This collapsing of code point levels may be detected based on the proximity of measured levels and, thus, measured levels which are sufficiently close in proximity are averaged irrespective of the actual code point associated with the measured levels. The increase in the number of samples may further reduce the impact of noise in the measured levels on subsequent utilization of the measured levels in, for example, constellation generation and detection of digital impairments such as digital pads and robbed-bit signaling.
Such proximity averaging may be provided by methods, systems and computer program products for averaging measured levels of code point sequences from a plurality of frame intervals in the presence of robbed-bit signaling. Levels of code points from the frame intervals of different code points are evaluated to determine if they should be averaged together by comparing a first measured level of a first code point from a first of the plurality of frame intervals and a measured level of a second code point from the plurality of frame intervals. The measured level of the second code point may be identified for incorporation into an average the measured levels which includes the first code point from the plurality of frame intervals based on the comparison of the first measured level of the first code point from the first of the plurality of frame intervals and the measured level of the second code point from the plurality of frame intervals and a threshold value. After the average is determined for a code point, the measured level of the second code point may then be replaced with the average of the measured levels of the first code points if the measured level of the second code point is incorporated into the average of the measured levels of the first code point. Preferably, the first and second code points are adjacent code points in the code point sequence. This process may be repeated for each measured level for each interval and then an average computed for all measured levels to be incorporated into the average. The replacement of the measured levels may then be performed using the resulting average.
It is also preferred that the measured level of the second code point be excluded from the average of the measured levels of the second code points from the plurality of frame intervals if the measured level of the second code point is incorporated into the average of the measured levels for the first code point. By excluding the measured level of the second code point from the average of the second code point, the second code point average will not be corrupted by a measured level which has been affected by robbed-bit signaling or other digital impairments which would cause the measured level to be collapse onto the value of an adjacent code point.
The average of the measured levels of the first code point may be established by comparing the first measured level of the first code point from the first of the plurality of frame intervals with a second measured level of the first code point from a second of the plurality of frame intervals. The second measured level of the first code point is then included in the average of the measured levels for the first code point from the plurality of frame intervals based on the comparison with the first measured level and a threshold value. After determining the average, the second measured level of the first code point may be replaced with the average of the measured levels of the first code points if the second measured level of the first code point is incorporated into the average of the measured levels of the first code point. This process may be repeated for each interval of the first code point so as to determine which measured levels are within the threshold with the first measured level. Thus, measured levels from intervals from the same code point level may only be averaged if they are sufficiently close in proximity to the first measured levels, which may exclude from an average within a code point those measured levels which have been affected by robbed-bit signaling to collapse onto another code point.
In a particular embodiment of the present invention, the measured level of the second code point is incorporated into an average of the measured levels for the first code point from the plurality of frame intervals if the first comparison result indicates that a difference between the measured level of the first code point and the measured level of the second code point is less than the threshold value.
Furthermore, because of the non-linear distribution of code point levels in mu-law and A-law PCM codes the threshold value for determining if a measured level is to be included in an average may be based on the measured levels of the code point sequence. Preferably, this is accomplished by determining an average spacing for measured levels of the code point sequence consecutive with the first code point and setting the threshold value based on the determined average spacing. In particular, the average spacing may be determined based on the difference between the first measured level of the first code point and a measured level of a third code point which is four code points lower in the code point sequence than the first code point.
In a specific embodiment of the present invention, the threshold value is established based on a selected measured level of the first code point. In such a case, the comparison of the measured level of the second code point value and the measured level of the first code point value may be accomplished by comparing the measured level of the second code point to the threshold value.
In particular embodiments of the present invention, the threshold value may be established by weighting the average spacing to provide a weighted average spacing and combining the weighted average spacing and the selected measured level of the first code point to provide the threshold value. If the selected measured level of the first code point is the smallest measured level of the first code point, then the threshold value may be established by weighting the average spacing to provide a weighted average spacing and adding the weighted average spacing and the selected measured level of the first code point to provide the threshold value. In such a case, the measured level of the second code point may be compared to the threshold value by determining if the measured level of the second code point is less than or equal to the threshold value.
If the selected measured level of the first code point is the largest measured level of the first code point, then the threshold value may be established by weighting the average spacing to provide a weighted average spacing and subtracting the weighted average spacing and the selected measured level of the first code point to provide the threshold value. In such a case, the measured level of the second code point may be compared to the threshold value by determining if the measured level of the second code point is greater than or equal to the threshold value.
As will be appreciated by those of skill in the art, the present invention may be embodied as methods, systems and/or computer program products.